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Unified Diff: webrtc/media/engine/webrtcvoiceengine.h

Issue 1728503002: Replace scoped_ptr with unique_ptr in webrtc/media/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@up1
Patch Set: Created 4 years, 10 months ago
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Index: webrtc/media/engine/webrtcvoiceengine.h
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index 55655e3887f20f9fa1fc587f9c7795b7fbe0e93a..dbb7ea6e76715a811b75a9b907c0e7c4b077fef1 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -12,12 +12,12 @@
#define WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
#include <map>
+#include <memory>
#include <string>
#include <vector>
#include "webrtc/audio_state.h"
#include "webrtc/base/buffer.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/stream.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/call.h"
@@ -113,7 +113,7 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback {
rtc::ThreadChecker worker_thread_checker_;
// The primary instance of WebRtc VoiceEngine.
- rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
+ std::unique_ptr<VoEWrapper> voe_wrapper_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
// The external audio device manager
webrtc::AudioDeviceModule* adm_ = nullptr;
@@ -188,7 +188,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
void SetRawAudioSink(
uint32_t ssrc,
- rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override;
+ std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
// implements Transport interface
bool SendRtp(const uint8_t* data,
@@ -240,7 +240,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
WebRtcVoiceEngine* const engine_ = nullptr;
std::vector<AudioCodec> recv_codecs_;
std::vector<AudioCodec> send_codecs_;
- rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
+ std::unique_ptr<webrtc::CodecInst> send_codec_;
bool send_bitrate_setting_ = false;
int send_bitrate_bps_ = 0;
AudioOptions options_;
@@ -258,7 +258,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
// Volume for unsignalled stream, which may be set before the stream exists.
double default_recv_volume_ = 1.0;
// Sink for unsignalled stream, which may be set before the stream exists.
- rtc::scoped_ptr<webrtc::AudioSinkInterface> default_sink_;
+ std::unique_ptr<webrtc::AudioSinkInterface> default_sink_;
// Default SSRC to use for RTCP receiver reports in case of no signaled
// send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
// and https://code.google.com/p/chromium/issues/detail?id=547661
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